This invention relates to the synchronization of digital signals, formed from successive words representing quantized samples of sound information and received from an asynchronous source, with a local reference clock; it is particularly--though not exclusively--suitable for use in the field of digital sound broadcasting.
The present techniques for transmitting, broadcasting and recording sound sources frequently involve coding the signals from such sources in digital form. The analog sound signal, supplied by the source, having a predetermined passband, is transformed into a succession of words of each x bits, called samples, representative of quantized values of the analog signal at specific regularly spaced times, called sampling times. The values of the sampling period and of the number x of bits per sample characterize the quality of the coding and depend on the nature of the signals to be processed. After transmission or recording, the digital data must be reconverted by a D/A converter which requires a local reference timing clock or time base. In order to generate the reference clock, the times of arrival of the successive samples are extracted from the received train of words. Their nominal frequency--i.e. the frequency determined by integration over a long period of time--is equal to the sampling frequency used for A/D conversion of the sound signal.
Unfortunately, the equipment through which the signal passes between the A/D and D/A conversions (line equipment in the case of transmission or broadcasting; and digital equipment in the case of recording) causes disturbances in the spectrum of the reconstituted clock rate. For example, the multiplexing equipment in digital networks is able to accept a limited amount of phase jitter, but it creates jitter which has low frequency spectral components which cannot be easily eliminated. Recorders exhibit similar deficiencies when they are not designed for servo-controlling the time base.
Moreover, it is often difficult or impractical to control a reception station for synchronizing it with an external timing rate, if the station has its own reference clock.
Multiplexing processes are already known for data transmission by "frames" in which "blank" words are inserted at a predetermined rate (French Patent No. 2,178,418). The result to be achieved is mixing of several incoming frames so as to form a single output signal. There is no processing of the signal. The article "A switching unit for integrated PCM communication", Conference on Switching Techniques for Telecommunication Networks, Apr. 21-25, 1969, pp. 157-160, describes a technique which involves prestuffing null words beforehand into the message. Again, the process does not use the particular features of the digital sound signals and is not applied to synchronization of a receiver which processes the signal.
In "A new digital voice actuated switch", Comsat Technical Review, Vol. 6, No. 1, pp. 159-178, there is described a procedure for detecting time intervals in a signal representing the voice when silence occurs. Such a procedure is specific to that type of signal and is used for time multiplexing only. British patent application No. 2,039,448 describes a method and apparatus for altering the timing of digital sound signal samples during substantially silent intervals which are detected by measuring the differences between successive adjacent samples or the absolute values of successive samples.
It is an object of the invention to synchronize received digital sound information signals with a local clock asynchronous with said signals, i.e. to match the timing of the digital signal to that of the clock, in a simple way; it is a more specific object to provide for efficient detection of "silent" intervals during which synchronization may be achieved by omitting or repeating samples representative of silences.
In this respect, it should be remembered that digital signals obtained from a sound source include silent periods, which may be lengthened or shortened without substantial drawback.
According to an aspect of the invention, a process for synchronizing digital signals consisting of successive words representing quantized samples of sound information and comprising silent periods with a local reference clock during conversion or restitution, comprises the steps of storing the words in a local buffer store as they are received and reading them at the frequency of a local clock. The degree of filling or occupancy of the store is continuously monitored and compared with a range between predetermined maximum and minimum filling thresholds. The contents of the store are modified upon detection of the first silent interval after it has been determined that the degree of occupancy is outside said range. For that purpose, the store is emptied by discording a number of samples if the degree of occupancy is higher than the maximum threshold, filled by repeating samples in the opposite case.
In other words, the degree of occupancy of the store is monitored continuously, after it has been filled up to a predetermined level (typically half its capacity) before the beginning of reading out. Depending on the difference between the writing (or input) rate and reading out (or output) rate, the store will fill up or empty. Above a predetermined upper filling threshold, the output flow will be artificially and temporarily increased during the first silent interval to occur. Under a lower filling threshold, the output flow will be artificially and temporarily reduced during the first silent interval.
The process of the invention involves searching for silent periods in the digital signal and an important aspect of the invention relates to such search. According to the invention, an interval will be considered as silent if corresponding to a continuous sequence of N samples (N being greater than a predetermined number N.sub.0) whose values are comprised between two predetermined extreme values S1 and S2. The maximum S1 and minimum S2 values may be implicit or programmable as explained in more detail hereunder. A possible DC offset of the sound signal during the A/D conversion may be corrected by appropriate programming of S1 and S2.
The parameter N.sub.0 characterizes the minimum time length during which the samples should be lower than thresholds for detection of a silence: its value may be implicit or programmable as well.
It can be seen that a silent interval is at least partially discarded when an increase of the output flow is desired: the stored contents then decrease by N samples if the silent interval is totally deleted; a silent interval may be repeated--or extended by a predetermined duration--when a reduction of the output flow is required for synchronization.
The value N.sub.0 may be selected depending on the relative permissible maximum difference between the writing and reading out frequencies; and on the statistical distribution of the "silent intervals" in the modulation as received.
A device according to the invention suitable for carrying out the above-defined process, comprises a random access memory stored associated with means for writing the words therein at the frequency of their arrival and with means for reading out the store at the frequency of a local clock; means for detecting the degree of filling of the store and for comparing it with range limited by predetermined minimum and maximum filling thresholds; a silent interval detector; and means controlled by the comparison means and the detector for partially filling or emptying the store, as the case may be, during the first silent interval after the degree of filling has passed outside the range.
In the above definitions, the words "sound information" are to be understood as designating any information having a time distribution comparable to that coming from a sound source properly speaking, comprising therefore silent intervals.
The invention will be better understood from the following description of an embodiment, given by way of example only.